Plugin Demo: NoSIP (SDP/RTP)

Demo details

This is a demo that complements the one showcasing the SIP plugin. In
fact, while the SIP plugin allows you to not worry about SIP details, which are implemented within
the plugin itself, the NoSIP plugin doesn't mess with signalling itself, leaving it up to the
application. As such, it provided an alternative to those that still want to interact with a legacy
infrastructure (e.g., a pre-existing SIP-based one), but still want to be able to have control
on the signalling themselves, rather than completely delegating it to the SIP plugin.

All this plugin does, as a consequence, is taking care of the translation between WebRTC
empowered SDPs, and barebone SDPs that can be used with legacy peers. The barebone SDPs the
plugin generates are crafted so that media is handled by the plugin itself, thus implementing
the same RTP/RTCP gateway functionality the SIP plugin provides, but without the constraint
of the signalling. It is up to the appplication to transport a generated offer in whatever
signalling they want to use (e.g., SIP, IAX, XMPP, etc.) and make sure the offer/answer from
the peer is passed to the plugin, so that the session can be completed.

Considering this plugin is very much generic and signalling-agnostic, this demo does NOT
involve any signalling at all. On the contrary, it will show how a WebRTC peer can establish
a session with another WebRTC peer (for the sake of simplicity located in the same page)
by passing through the RTP/RTCP gatewaying functionality. This should as a result make it easier
for you to understand how a NoSIP caller and a NoSIP callee would need to be implemented. The
barebone SDPs generated/processed as a consequence will be displayed as a proof of concept.